[asterisk-commits] file: branch 1.4 r105674 - in /branches/1.4: channels/ include/asterisk/ main/

[SVN commits to the Asterisk project]

Author: file
Date: Tue Mar  4 12:05:28 2008
New Revision: 105674

URL: http://svn.digium.com/view/asterisk?view=rev&rev=105674
Log:
When a new source of audio comes in (such as music on hold) make sure the marker bit gets set.
(closes issue #10355)
Reported by: wdecarne
Patches:
      10355.diff uploaded by file (license 11)
(closes issue #11491)
Reported by: kanderson

Modified:
    branches/1.4/channels/chan_sip.c
    branches/1.4/include/asterisk/rtp.h
    branches/1.4/main/rtp.c

Modified: branches/1.4/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=105674&r1=105673&r2=105674
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Tue Mar  4 12:05:28 2008
@@ -3661,8 +3661,10 @@
 			if (option_debug > 1)
 				ast_log(LOG_DEBUG,"T38State change to %d on channel %s\n", p->t38.state, ast->name);
 			res = transmit_response_with_t38_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL);
-		} else 
+		} else {
+			ast_rtp_new_source(p->rtp);
 			res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL);
+		}
 	}
 	ast_mutex_unlock(&p->lock);
 	return res;
@@ -3695,6 +3697,7 @@
 				if ((ast->_state != AST_STATE_UP) &&
 				    !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
 				    !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
+					ast_rtp_new_source(p->rtp);
 					transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE);
 					ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);	
 				}
@@ -3920,9 +3923,11 @@
 		res = -1;
 		break;
 	case AST_CONTROL_HOLD:
+		ast_rtp_new_source(p->rtp);
 		ast_moh_start(ast, data, p->mohinterpret);
 		break;
 	case AST_CONTROL_UNHOLD:
+		ast_rtp_new_source(p->rtp);
 		ast_moh_stop(ast);
 		break;
 	case AST_CONTROL_VIDUPDATE:	/* Request a video frame update */

Modified: branches/1.4/include/asterisk/rtp.h
URL: http://svn.digium.com/view/asterisk/branches/1.4/include/asterisk/rtp.h?view=diff&rev=105674&r1=105673&r2=105674
==============================================================================
--- branches/1.4/include/asterisk/rtp.h (original)
+++ branches/1.4/include/asterisk/rtp.h Tue Mar  4 12:05:28 2008
@@ -161,6 +161,8 @@
 int ast_rtp_sendcng(struct ast_rtp *rtp, int level);
 
 int ast_rtp_settos(struct ast_rtp *rtp, int tos);
+
+void ast_rtp_new_source(struct ast_rtp *rtp);
 
 /*! \brief  Setting RTP payload types from lines in a SDP description: */
 void ast_rtp_pt_clear(struct ast_rtp* rtp);

Modified: branches/1.4/main/rtp.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/main/rtp.c?view=diff&rev=105674&r1=105673&r2=105674
==============================================================================
--- branches/1.4/main/rtp.c (original)
+++ branches/1.4/main/rtp.c Tue Mar  4 12:05:28 2008
@@ -172,6 +172,7 @@
 	struct ast_rtcp *rtcp;
 	struct ast_codec_pref pref;
 	struct ast_rtp *bridged;        /*!< Who we are Packet bridged to */
+	int set_marker_bit:1;           /*!< Whether to set the marker bit or not */
 };
 
 /* Forward declarations */
@@ -1994,6 +1995,12 @@
 	return res;
 }
 
+void ast_rtp_new_source(struct ast_rtp *rtp)
+{
+	rtp->set_marker_bit = 1;
+	return;
+}
+
 void ast_rtp_set_peer(struct ast_rtp *rtp, struct sockaddr_in *them)
 {
 	rtp->them.sin_port = them->sin_port;
@@ -2641,6 +2648,13 @@
 			}
 		}
 	}
+
+	/* If we have been explicitly told to set the marker bit do so */
+	if (rtp->set_marker_bit) {
+		mark = 1;
+		rtp->set_marker_bit = 0;
+	}
+
 	/* If the timestamp for non-digit packets has moved beyond the timestamp
 	   for digits, update the digit timestamp.
 	*/